What is SIP?
Telephone voice communication over the Internet between 2 and several subscribers requires a transmission protocol. The modern world of Internet telephony speaks in this context of Session Initiation Protocol - SIP in short, which has already established itself as the standard for voice-over-IP (voice over the Internet Protocol). SIP regulates the context of the session in connection with voice transmission.
Although further protocols are required for complete telephony over the Internet (such as SDP - Session Description Protocol and RTP - Real-Time Transport Protocol), 99% of the people involved use the term SIP to describe the network protocol used for the setup, the control and the dismantling of a communication session.
So in summary, SIP is THE technology that brings the standard of Voice over Internet Protocol (VoIP) to modern businesses.
What are codecs?
Basically, a codec refers to an algorithm pair that digitally encodes and decodes data or signals. As part of a data exchange, the codecs used by the participating devices are compared to ensure that they "understand" each other.
What does SIP have to do with codecs?
Communicating devices share in the course of the SIP session with each other which methods of video and audio transmission (codecs) are known to them and can be used. This avoids data being incorrectly decoded and not readable.
What are the advantages of SIP?
- SIP is an open standard, which has become widely used due to its flexibility
- SIP servers are distributed and thus more difficult to attack
- already established sessions can be modified
What is a SIP trunk?
A trunk basically stands for a (telephone) line. With a SIP Trunk, the call is initiated, controlled and cleared via the Session Initiation Protocol through the Internet. SIP trunks replace the previous (ISDN-, PSTN- and other) telephone line trunks to allow improved communication to both IP networks and legacy (ISDN or legacy) systems. A SIP trunk can be connected to your IP-capable telephone system or a softswitch. 1 SIP trunk (= 1 SIP account) is usually assigned to 1 main call number - the number of attached extensions is flexible. This SIP account allows the direct dialing to many terminals, whereby the administration of the branches takes place completely in the IP-able telephone system (IP PBX).
It should be noted that 1 SIP trunk offers several voice channels - the number of possible channels depends on the bandwidth of the IP connection and the settings of the telephone system and the provider.
What is SIP Trunking?
SIP Trunking is simply the use of SIP trunks, in other words: SIP Trunking is the process by which SIP technology is applied to VoIP systems.
Why you need SIP trunks?
In order to benefit from the advantages of modern Internet telephony (cost reduction, future security, location independence, additional functions, ...), you need one or more SIP trunks instead of the old ISDN or PSTN telephone lines.
Who offers SIP Trunks?
Most telecom providers already offer their customers SIP trunks. VoIP providers (such as www.mediatel.at) usually offer only SIP Trunks as connection variants. Find out exactly what bandwidth the trunks of your provider offers.
What should I pay attention to when selecting the SIP trunk provider?
Find a reliable, low-cost provider that provides reasonable bandwidth to run many simultaneous voice channels in 1 trunk. Pay attention to the scalability of your SIP Trunk - it should be possible flexibly to add further extensions depending on business and corporate development. We also recommend that you inform yourself in advance about the number porting.
Which telephone systems does media.tel SIP support?
Our customers use a wide range of equipment. If you do not find your manufacturer in the list below, then let us know:
Sangoma, Elastix, 3CX, Aastra, Auerswald, Alcatel, Asterisk, Avaya, Cisco, Gigaset, Nortel, Panasonic, Siemens, Starface, etc.
Thanks to https://unsplash.com/@rawpixel for the cool image.